Internet protocol (IP) based communication systems allow the user of a device, such as a personal computer, to make calls across a computer network such as the Internet. These systems are beneficial to the user as they are often of significantly lower cost than traditional telephony networks, such as fixed line or mobile networks. This may particularly be the case for long distance voice calls. These systems may utilise voice over internet protocol (“VoIP”) over an existing network (e.g. the Internet) to provide these services, although alternative protocols can also be used. These systems may also provide further services such as video calling and instant messaging (“IM”).
Two main types of IP-based communication systems operate today. The first operates using a peer-to-peer (“P2P”) topology built on proprietary protocols. An example of this type of communication system is the Skype™ system. To use a peer-to-peer service, the user must install and execute client software on their PC, and register with the P2P system. When the user registers with the P2P system the client software is provided with a digital certificate from a central server. Once the client software has been provided with the certificate calls can subsequently be set up and routed between users of the P2P without the further use of a central server. The client software provides the VoIP and IM connections. It is therefore a characteristic of peer-to-peer communication that the call is not routed using the central server but directly from end-user to end-user. Further details on such a P2P system are disclosed in WO 2005/009019.
The second main type of IP-based communication system is based on the session initiation protocol (“SIP”). SIP is a communications signalling protocol developed by the Internet Engineering Task Force (“IETF”) and is a proposed standard for initiating, modifying and terminating an interactive user session that involves multimedia elements such as voice, video and IM. SIP is the leading signalling protocol for VoIP. SIP is described in IETF RFC 3261. A user of a SIP service is generally provided with either a telephone number (of the known e.164 format) or a SIP uniform resource identifier (“URI”) (of the form SIP:username@example.com). To call a SIP user, the caller uses the SIP:URI or e.164 number to identify the SIP user, and this is translated to the IP addresses of the called user's terminal by a registrar database.